NOTE: Please make sure that you use only the genuine batteries in your handset and they are inserted properly!

If you are intending to use PSTN, please connect your phone this way

1. Ordinary telephone wall-plug2.Gigaset C 455 IP Base module.

2.1.Wallmountrabbets2.2. Power jack2.3. Phone jack

3. Electric wall-plug

The PSTN(2.1.) and POWER (2.2.) connectors are located at the bottom of the Base module(2).

If not using PSTN please connect your IP phone following this hint:

* Connect the ethernet cable delivered in your Gigaset C 455 IP package to the network switch or the wall-plug(3) and insert its free end in the LAN jack(2) on your Gigaset C 455 IP Base module(1).

* Connect your PC to the network switch using another ethernet cable. * Connect the Internet cable to the network switch .* Connect the power cable to the Gigaset C 455 IP Base Module and to the appropriate wall-plug.

NOTE: Please use only the mains adapter supplied!

When all the cables are connected, it is time to pair your Handset and the Base module.
It should be registered by default.

A successfully registered handset has the „INT“ display key on the bottom left side of its display.

If your handset does not register automatically please follow these steps:

Here you can define your dialing connection.
It could be a custom IP connection
You can use a Gigaset.net account, if you have one;
You can use a fixed line too.

You can edit an IP connection to enter your own details by clicking the appropriate „edit“ button on the page.

Then you will access the configuration screen

You can enter your connection's name or number here;

NOTE: If you do not use local VoIP provider account, please skip to the next paragraph!

choose a VoIPprovider if there is such in your country AND you have a working account by pressing

You will access the next page

click „NEXT“ to get the list of available providers list.

If you will use different provider, as for example your own Asterisk server, please enter your authenticationname, password, username and Displayname

NOTE: the authentication name and the authentication password are the same you have in your Asterisk Configuration files! Please read below to learn how to create a SIP account on your Asterisk server.

Choose if you will use Call Forwarding and in what circumstances, and number the incoming calls should be forwarded to.

To configure the VoIP server network settings please use the

button.

You will see the page extends downwards:

Here you can enter your Asterisk server's domain, IPaddress, choose if you need to use a Proxy or a STUN server if you deal with devices behind NAT, ports, Volumesettings, audiocodecs and etc.

Number Assignment

If you have more than one lines and more than one handset, you can bind each of the handsets to different line.
Here you can bind your Answeringmachine to a selected line.

Dialing Plans

If you will make local calls through your IPphone, please enter your Areacode here.
You can enter some emergency numbers here too if you use a FixedLine.

Directory

Set your handset directory here.

Advanced Settings

Here you can set such important things as RTPport or SIPport, DTMF settings and etc.

„Messaging“

Here you have another submenu with two choices

Messenger

Enter here your messenger details to send and receive messages.

Email

Enter your email settings here if you want to use it with your IP phone

„Miscellaneous“

Here you can adjust the firmware update settings – by using given server, or file, and update the running Firmware by clicking the „UpdateFirmwareVersion“ button.

Status

Here you can see the current status of your device.

To confirm all the changes, please click the „SET“ button and reboot your device.

Setting up an Asterisk account

Setting up an Asterisk account to be used with your hardphone.

NOTE: If you are an advanced Asterisk user, please skip the following paragraph about setting up an Asterisk account to use with your phone.

To set up a working SIP account on your Asterisk server, you have to edit some .conf files. Please, connect using your mostly preferred way to your Asterisk server and find where the sip.conf and extensions.conf files are located. They are usually located at the /etc/asterisk/ directory. Now, use your favorite text editor to make some changes. First edit the sip.conf file.
Adding a new user is a simple process, which consists of adding some lines, similar to these in our example here:

I added the following to my sip.conf :

[user]: needed to indicate we are registering user "user". username=user: to determine our user's name is "user". context=conf: defines the dial context for the user "user". Asterisk divides outgoing numbers in groups called contexts in order to separate/define different needs for different uses. In this example the user "user" is in a context called "conf".type=friend: Users that can place AND receive calls are described as "friend". If your new user will only receive calls, use "peer" as type. For placing calls only use "user" as type. secret=123: this is where you set up your user's password. He will use it to login/authenticate on Asterisk. It is presented in plain text, bus as only the server's administrators (you) have access to this file, there is nothing to worry about. allow=all: means that the line which this user will use supports all audio codecs. host=dynamic: You can define your users IP as static or dynamic, according to the type of network. rfc2833compensate=yes: You must have this turned on, or DTMF reception will work improperly.nat=yes: If you are placing the phone behind a NAT you must enable that.

Save the changes and continue to register an extension for the user "user" ;).

Now please open the file extensions.conf in your favorite editor to make some changes:

In my file I added :

conf: it is the name of the new context.
The first line "exten => 731,1,Dial(SIP/user)" shows that if somebody dials the number 731 his call will be connected with the user "user" through SIP.
The second line "exten => 731,2,Hangup()" says that if the conversation is over then Asterisk has to hang up the line. It is needed to be sure that Asterisk will free it for the next call.

NOTE: Do not forget to save the changes you just made in extensions.conf .

To know that everything is all right you can test it with an echo test, or by installing a softphone.

Echo test

To do an echo test you need to add few more lines in your extensions.conf file.
Open it with your favorite editor and edit it following the example:

exten => 111,1,Answer(): that means that the call will be answered;exten => 111,2,Echo(): makes possible for you to hear your own voice;exten => 111,3,HangUp(): hangs up after you've finished the call.

Now, if you dial 111 and say something you will hear your own voice.

Testing by a softphone

To make a test call with a softphone you need one.
Download Zoiper and test it for free from the download section on our website!

Installation

Zoiper on Linux

If you are using Linux, please follow these steps which will lead you to a working Zoiper phone.

1. Download the appropriate archive from our website;
2. choose a destination directory and extract it there;
3. open that directory and click the newly extracted file Zoiper with your mouse to run it. You can also open a terminal in the active X-session, navigate to the target directory by using cd and execute the file by

# ./zoiper

Zoiper on Windows

If you are using Windows, please follow these steps which will lead you to a working Zoiper phone.

1. Download the appropriate archive from our website;
2. run the newly downloaded .exe file;
3. you will see Zoiper's installation wizard. You will be expected to read a License Agreement, and maybe - agree it.
4. If you agree it, please click on the appropriate button to proceed to the next screen, to choose a destination directory for your Zoiper. You can see some information about space required by Zoiper and the available free space on your hard drive.
5. The next screen is an opportunity to choose where do you want your Zoiper Start Menu shortcuts.
6. Once again you will deal with shortcuts – this time some additional ones – for your Desktop and Quick Launch. If you do not want them, just unmark the appropriate checkboxes.
7. When you click the “Install” button the real installation will begin. You can click the “Details button to see some details, but it will finish so fast that you'd hardly have enough time.
8. The next screen is the last one. It will inform you that your Zoiper installation had finished successfully.

Congratulations ! You can start using it now !.

Setting an additional account.

Please connect/open a terminal session to your Asterisk server in order to modify your /etc/asterisk/sip.conf file and add one more user account. It will be needed to use with Zoiper.

You can copy the “user” account, and modify it a little.

[user]:change it to [user2] to determine that we are setting an account for “user2”username=user: change it too -&gt; username=user2to determine our user's name is now "user2".context=conf: defines the dial context for the user "user2". You should leave it as it is, as “user” and "user2" can be in the same context.type=friend: leave as it is, as you will need both of your users to place and receive calls when testing.secret=123: you may use the same password or change it.allow=all: you do not need to change that.host=dynamic: you can either enter the IP address of the computer you've just installed Zoiper on or leave this as it is.rfc2833compensate=yes: You must have this turned on, or DTMF reception will work improperly.nat=yes: If you are placing the phone behind a NAT you must enable that.

Now save this file and open /etc/extensions.conf to add a new extension for that user.

Setting additional extensions.

You added 3 lines for your first account. Now add the new ones below:

As adding user 2 to the existing context, you do not need to specify a new one.
Just add these two lines:

exten => 732,1,Dial(SIP/user2) shows that if somebody dials the number 732 his call will be connected with the user "user2" through SIP.exten => 732,2,Hangup(): it is needed to be sure that when the conversation is over then Asterisk has to hang up the line. It is needed to be sure that Asterisk will free it for the next call.

Zoiper Setup

To use your new Zoiper softphone you need an account and you have to set it up.
This you can do following my steps:

Accessing Options Form

Start your Zoiper and right-click on its interface. Click on "Options" to open the configuration menu.

There is an Options button on the Zoiper’s interface. You can start configuration form from there too.

Use "Alt+O" if you prefer using keyboard than clicking.

Adding SIP account

When the form starts you should click on the "Add new SIP account" label in the navigation menu to the left.

A new sub-form will appear. Enter a name the account and click on the "OK" button to create it.
The new entry will appear in the navigation menu to the left.
Click on it to continue the configuration.

On the right side of the navigation menu you can specify some options for the account.
Enter the hostname or the IP address of the Asterisk server (or other VoIP server) that you are going to use.
Enter the account name and secret that you put in the sip.conf file (user2/012). Enter a caller id and name and click on the "Apply" button to activate the changes.

There are additional options that you might want to configure. In order to view the advanced options for the current account you should enable the checkbox with label "Show advanced options" which is located on the bottom left corner.

When you do this you can set an outbound proxy, if necessary, or a voicemail extension, a STUN server, if needed. You can select if Zoiper should register this account upon application’s start-up.

When you are ready click on the "Apply" button.

Registering SIP accounts

You should select the proper account from the drop down menu in the bottom of Zoiper’s interface. When you are ready with this click on the "Register" button.

When you reset your Base all settings are reset.
The system PIN will be reset to "0000" and all handsets are deregistered.

* Remove the cable connections from the base station to the router and fixed network.
* Remove the base station mains unit from the socket.
* Press and hold the registration/paging key.
* Plug the mains unit back into the power socket.
* Press and hold the registration/paging key for at least 10 seconds.
* Release the registration/paging key.

Andrei (amaksimov at tochka dot ru)16 March 2008 22:42:17My S675 IP prompted me that there was a new firmware update, I hit the button to install it, and it snot worked since. Tried resetting it, tried getting via the web access, the handsets just won't see the base station! Please help!!!

javame (vacula dot jan at gmail dot com)12 February 2008 09:48:39Hello, I have some problem with this nice phone. I need to use it in the plant, where is possible only calling of IP address, not number extensions. From C450 I can call IP of all phones, but when I want to call IP of C450, others phones said it is busy. Has anybody the same problem? Would anybody know the solution?

(sorry for my english, I've never been good student)

Add Comment

Name:

Email:

Comment:

In order to prevent automatic posting on our website, we kindly request you to type in the number you see in the picture below.